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gstreamer/gst/rtsp-server/rtsp-media.h

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Initial import
2008-10-09 12:29:12 +00:00
/* GStreamer
* Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
Fix FSF address
2012-11-04 00:14:25 +00:00
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
Initial import
2008-10-09 12:29:12 +00:00
*/
#include <gst/gst.h>
Cleanup of sessions and more Fix the refcounting of media and sessions in the client. Properly clean up the session data when the client performs a teardown. Add Server header to responses. Allow for multiple uri setups in one session. Add Range header to the PLAY response and add the range attribute to the SDP message. Fix the session pool remove method, it used the wrong key in the hashtable. Also give the ownership of the sessionid to the session object.
2009-02-04 16:00:42 +00:00
#include <gst/rtsp/gstrtsprange.h>
Initial import
2008-10-09 12:29:12 +00:00
#include <gst/rtsp/gstrtspurl.h>
#ifndef __GST_RTSP_MEDIA_H__
#define __GST_RTSP_MEDIA_H__
G_BEGIN_DECLS
More improvements Rename GstRTSPMediaBin to GstRTSPMedia Parse the request url into a GstRTSPUri object and pass this object to the various handlers and methods that require the uri.
2009-01-22 16:58:19 +00:00
/* types for the media */
#define GST_TYPE_RTSP_MEDIA (gst_rtsp_media_get_type ())
#define GST_IS_RTSP_MEDIA(obj) (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_MEDIA))
#define GST_IS_RTSP_MEDIA_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_MEDIA))
#define GST_RTSP_MEDIA_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_MEDIA, GstRTSPMediaClass))
#define GST_RTSP_MEDIA(obj) (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_MEDIA, GstRTSPMedia))
#define GST_RTSP_MEDIA_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_MEDIA, GstRTSPMediaClass))
#define GST_RTSP_MEDIA_CAST(obj) ((GstRTSPMedia*)(obj))
#define GST_RTSP_MEDIA_CLASS_CAST(klass) ((GstRTSPMediaClass*)(klass))
Initial import
2008-10-09 12:29:12 +00:00
More improvements Rename GstRTSPMediaBin to GstRTSPMedia Parse the request url into a GstRTSPUri object and pass this object to the various handlers and methods that require the uri.
2009-01-22 16:58:19 +00:00
typedef struct _GstRTSPMedia GstRTSPMedia;
typedef struct _GstRTSPMediaClass GstRTSPMediaClass;
Initial import
2008-10-09 12:29:12 +00:00
rtsp: massive refactoring Make GObjects from the remaining simple structures. Remove GstRTSPSessionStream, it's not needed. Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how a GstRTSPStream should be transported to a client. Rename GstRTSPMediaFactory::get_element -> create_element because that more accurately describes what it does. Make nice methods instead of poking in the structures. Move some methods inside the relevant object source code. Use GPtrArray to store objects instead of plain arrays, it is more natural and allows us to more easily clean up. Move the allocation of udp ports to the Stream object. The Stream object contains the elements needed to stream the media to a client. Improve the prepare and unprepare methods. Unprepare should now undo everything prepare did. Improve also async unprepare when doing EOS on shutdown. Make sure we always unprepare correctly.
2012-10-25 19:29:58 +00:00
#include "rtsp-stream.h"
media: enable per factory authorisations Allow for adding a GstRTSPAuth on the factory and media level and check permissions when accessing the factory. Add hints to the auth methods for future more fine grained authorisation. Add example application for per factory authentication.
2011-01-12 12:57:09 +00:00
#include "rtsp-auth.h"
rtsp: use AddressPool Remove the multicast_group property. Use the configured addresspool to allocate multicast addresses.
2012-11-14 16:23:59 +00:00
#include "rtsp-address-pool.h"
media: enable per factory authorisations Allow for adding a GstRTSPAuth on the factory and media level and check permissions when accessing the factory. Add hints to the auth methods for future more fine grained authorisation. Add example application for per factory authentication.
2011-01-12 12:57:09 +00:00
media: avoid doing _get_state() for state changes When preparing, use the ASYNC_DONE and ERROR messages in the bus handler to wait until the media is prerolled or in error. This avoids doing a blocking call of gst_element_get_state() that can cause lockups when there is an error. Fixes #611899
2010-03-05 16:51:26 +00:00
/**
* GstRTSPMediaStatus:
* @GST_RTSP_MEDIA_STATUS_UNPREPARED: media pipeline not prerolled
rtsp: massive refactoring Make GObjects from the remaining simple structures. Remove GstRTSPSessionStream, it's not needed. Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how a GstRTSPStream should be transported to a client. Rename GstRTSPMediaFactory::get_element -> create_element because that more accurately describes what it does. Make nice methods instead of poking in the structures. Move some methods inside the relevant object source code. Use GPtrArray to store objects instead of plain arrays, it is more natural and allows us to more easily clean up. Move the allocation of udp ports to the Stream object. The Stream object contains the elements needed to stream the media to a client. Improve the prepare and unprepare methods. Unprepare should now undo everything prepare did. Improve also async unprepare when doing EOS on shutdown. Make sure we always unprepare correctly.
2012-10-25 19:29:58 +00:00
* @GST_RTSP_MEDIA_STATUS_UNPREPARING: media pipeline is busy doing a clean
* shutdown.
media: avoid doing _get_state() for state changes When preparing, use the ASYNC_DONE and ERROR messages in the bus handler to wait until the media is prerolled or in error. This avoids doing a blocking call of gst_element_get_state() that can cause lockups when there is an error. Fixes #611899
2010-03-05 16:51:26 +00:00
* @GST_RTSP_MEDIA_STATUS_PREPARING: media pipeline is prerolling
* @GST_RTSP_MEDIA_STATUS_PREPARED: media pipeline is prerolled
* @GST_RTSP_MEDIA_STATUS_ERROR: media pipeline is in error
*
* The state of the media pipeline.
*/
typedef enum {
rtsp: massive refactoring Make GObjects from the remaining simple structures. Remove GstRTSPSessionStream, it's not needed. Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how a GstRTSPStream should be transported to a client. Rename GstRTSPMediaFactory::get_element -> create_element because that more accurately describes what it does. Make nice methods instead of poking in the structures. Move some methods inside the relevant object source code. Use GPtrArray to store objects instead of plain arrays, it is more natural and allows us to more easily clean up. Move the allocation of udp ports to the Stream object. The Stream object contains the elements needed to stream the media to a client. Improve the prepare and unprepare methods. Unprepare should now undo everything prepare did. Improve also async unprepare when doing EOS on shutdown. Make sure we always unprepare correctly.
2012-10-25 19:29:58 +00:00
GST_RTSP_MEDIA_STATUS_UNPREPARED = 0,
GST_RTSP_MEDIA_STATUS_UNPREPARING = 1,
GST_RTSP_MEDIA_STATUS_PREPARING = 2,
GST_RTSP_MEDIA_STATUS_PREPARED = 3,
GST_RTSP_MEDIA_STATUS_ERROR = 4
media: avoid doing _get_state() for state changes When preparing, use the ASYNC_DONE and ERROR messages in the bus handler to wait until the media is prerolled or in error. This avoids doing a blocking call of gst_element_get_state() that can cause lockups when there is an error. Fixes #611899
2010-03-05 16:51:26 +00:00
} GstRTSPMediaStatus;
Add comments. Remove unused field
2009-01-20 18:43:47 +00:00
/**
More improvements Rename GstRTSPMediaBin to GstRTSPMedia Parse the request url into a GstRTSPUri object and pass this object to the various handlers and methods that require the uri.
2009-01-22 16:58:19 +00:00
* GstRTSPMedia:
docs: update docs
2012-10-26 10:33:21 +00:00
* @parent: parent GObject
media: update comments a little
2010-03-09 12:44:20 +00:00
* @lock: for protecting the object
* @cond: for signaling the object
Add shared properties to media and factory Add the shared property to media. Implement some simple caching in the factory depending on if the media is shared or not.
2009-01-29 16:20:27 +00:00
* @shared: if this media can be shared between clients
media: more work on making the media shared Add a reusable flag to medias, indicating that they can be reused after a state change to NULL. Small cleanups.
2009-04-03 20:22:30 +00:00
* @reusable: if this media can be reused after an unprepare
media: allow configuration of allowed lower transport
2010-03-19 14:15:29 +00:00
* @protocols: the allowed lower transport for this stream
media: update comments a little
2010-03-09 12:44:20 +00:00
* @reused: if this media has been reused
media: allow configuration of allowed lower transport
2010-03-19 14:15:29 +00:00
* @is_ipv6: if this media is using ipv6
docs: update docs
2012-10-26 10:33:21 +00:00
* @eos_shutdown: if EOS should be sent on shutdown
* @buffer_size: The UDP buffer size
* @auth: the authentication service in use
* @multicast_group: the multicast group to use
Reorganize things, prepare for media sharing Added various other test server examples Move the SDP message generation to a separate helper. Refactor common code for finding the session. Add content-base for realplayer compatibility Clean up request uris before processing for better vlc compatibility. Move prerolling and pipeline construction to the RTSPMedia object. Use multiudpsink for future pipeline reuse.
2009-01-29 12:31:27 +00:00
* @element: the data providing element
rtsp: massive refactoring Make GObjects from the remaining simple structures. Remove GstRTSPSessionStream, it's not needed. Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how a GstRTSPStream should be transported to a client. Rename GstRTSPMediaFactory::get_element -> create_element because that more accurately describes what it does. Make nice methods instead of poking in the structures. Move some methods inside the relevant object source code. Use GPtrArray to store objects instead of plain arrays, it is more natural and allows us to more easily clean up. Move the allocation of udp ports to the Stream object. The Stream object contains the elements needed to stream the media to a client. Improve the prepare and unprepare methods. Unprepare should now undo everything prepare did. Improve also async unprepare when doing EOS on shutdown. Make sure we always unprepare correctly.
2012-10-25 19:29:58 +00:00
* @streams: the different #GstRTSPStream provided by @element
media: update comments a little
2010-03-09 12:44:20 +00:00
* @dynamic: list of dynamic elements managed by @element
media: avoid doing _get_state() for state changes When preparing, use the ASYNC_DONE and ERROR messages in the bus handler to wait until the media is prerolled or in error. This avoids doing a blocking call of gst_element_get_state() that can cause lockups when there is an error. Fixes #611899
2010-03-05 16:51:26 +00:00
* @status: the status of the media pipeline
rtsp: massive refactoring Make GObjects from the remaining simple structures. Remove GstRTSPSessionStream, it's not needed. Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how a GstRTSPStream should be transported to a client. Rename GstRTSPMediaFactory::get_element -> create_element because that more accurately describes what it does. Make nice methods instead of poking in the structures. Move some methods inside the relevant object source code. Use GPtrArray to store objects instead of plain arrays, it is more natural and allows us to more easily clean up. Move the allocation of udp ports to the Stream object. The Stream object contains the elements needed to stream the media to a client. Improve the prepare and unprepare methods. Unprepare should now undo everything prepare did. Improve also async unprepare when doing EOS on shutdown. Make sure we always unprepare correctly.
2012-10-25 19:29:58 +00:00
* @n_active: the number of active connections
docs: update docs
2012-10-26 10:33:21 +00:00
* @adding: when elements are added to the pipeline
Reorganize things, prepare for media sharing Added various other test server examples Move the SDP message generation to a separate helper. Refactor common code for finding the session. Add content-base for realplayer compatibility Clean up request uris before processing for better vlc compatibility. Move prerolling and pipeline construction to the RTSPMedia object. Use multiudpsink for future pipeline reuse.
2009-01-29 12:31:27 +00:00
* @pipeline: the toplevel pipeline
media: update comments a little
2010-03-09 12:44:20 +00:00
* @fakesink: for making state changes async
Add TCP transports Use appsrc and appsink to send and receive RTP/RTCP packets in the TCP connection.
2009-03-11 15:45:12 +00:00
* @source: the bus watch for pipeline messages.
* @id: the id of the watch
* @is_live: if the pipeline is live
media: add a seekable boolean Maintain the seekable state with a new variable instead of reusing the is_live variable.
2011-11-03 11:55:24 +00:00
* @seekable: if the pipeline can perform a seek
Add TCP transports Use appsrc and appsink to send and receive RTP/RTCP packets in the TCP connection.
2009-03-11 15:45:12 +00:00
* @buffering: if the pipeline is buffering
* @target_state: the desired target state of the pipeline
Reorganize things, prepare for media sharing Added various other test server examples Move the SDP message generation to a separate helper. Refactor common code for finding the session. Add content-base for realplayer compatibility Clean up request uris before processing for better vlc compatibility. Move prerolling and pipeline construction to the RTSPMedia object. Use multiudpsink for future pipeline reuse.
2009-01-29 12:31:27 +00:00
* @rtpbin: the rtpbin
Add TCP transports Use appsrc and appsink to send and receive RTP/RTCP packets in the TCP connection.
2009-03-11 15:45:12 +00:00
* @range: the range of the media being streamed
media: support more Range formats Use the new -base methods to convert the Range string into a seek start and stop value.
2012-11-21 15:41:56 +00:00
* @range_start: range start in #GstClockTime
* @range_stop: range stop in #GstClockTime
Add comments. Remove unused field
2009-01-20 18:43:47 +00:00
*
Cleanups and doc updates Add some more documentation and do some minor cleanups here and there.
2009-01-22 17:35:17 +00:00
* A class that contains the GStreamer element along with a list of
rtsp: massive refactoring Make GObjects from the remaining simple structures. Remove GstRTSPSessionStream, it's not needed. Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how a GstRTSPStream should be transported to a client. Rename GstRTSPMediaFactory::get_element -> create_element because that more accurately describes what it does. Make nice methods instead of poking in the structures. Move some methods inside the relevant object source code. Use GPtrArray to store objects instead of plain arrays, it is more natural and allows us to more easily clean up. Move the allocation of udp ports to the Stream object. The Stream object contains the elements needed to stream the media to a client. Improve the prepare and unprepare methods. Unprepare should now undo everything prepare did. Improve also async unprepare when doing EOS on shutdown. Make sure we always unprepare correctly.
2012-10-25 19:29:58 +00:00
* #GstRTSPStream objects that can produce data.
Cleanups and doc updates Add some more documentation and do some minor cleanups here and there.
2009-01-22 17:35:17 +00:00
*
* This object is usually created from a #GstRTSPMediaFactory.
Add comments. Remove unused field
2009-01-20 18:43:47 +00:00
*/
More improvements Rename GstRTSPMediaBin to GstRTSPMedia Parse the request url into a GstRTSPUri object and pass this object to the various handlers and methods that require the uri.
2009-01-22 16:58:19 +00:00
struct _GstRTSPMedia {
media: avoid doing _get_state() for state changes When preparing, use the ASYNC_DONE and ERROR messages in the bus handler to wait until the media is prerolled or in error. This avoids doing a blocking call of gst_element_get_state() that can cause lockups when there is an error. Fixes #611899
2010-03-05 16:51:26 +00:00
GObject parent;
Initial import
2008-10-09 12:29:12 +00:00
rtsp-server: port to new thread API
2012-05-11 07:42:47 +00:00
GMutex lock;
GCond cond;
Add shared properties to media and factory Add the shared property to media. Implement some simple caching in the factory depending on if the media is shared or not.
2009-01-29 16:20:27 +00:00
media: avoid doing _get_state() for state changes When preparing, use the ASYNC_DONE and ERROR messages in the bus handler to wait until the media is prerolled or in error. This avoids doing a blocking call of gst_element_get_state() that can cause lockups when there is an error. Fixes #611899
2010-03-05 16:51:26 +00:00
gboolean shared;
gboolean reusable;
media: allow configuration of allowed lower transport
2010-03-19 14:15:29 +00:00
GstRTSPLowerTrans protocols;
media: avoid doing _get_state() for state changes When preparing, use the ASYNC_DONE and ERROR messages in the bus handler to wait until the media is prerolled or in error. This avoids doing a blocking call of gst_element_get_state() that can cause lockups when there is an error. Fixes #611899
2010-03-05 16:51:26 +00:00
gboolean reused;
rtsp: keep track of server ip and ipv6 Keep track of how the client connected to the server and setup the udp ports with the same protocol. Copy the server ip address in the SDP so that clients can send RTCP back to us.
2010-03-16 17:37:18 +00:00
gboolean is_ipv6;
media-factory: add eos-shutdown property Add an eos-shutdown property that will send an EOS to the pipeline before shutting it down. This allows for nice cleanup in case of a muxer. Fixes #625597
2010-08-20 16:17:08 +00:00
gboolean eos_shutdown;
media: add property to configure kernel buffer sizes Add a property to configure the kernel UDP buffer size.
2011-01-31 16:28:22 +00:00
guint buffer_size;
media: enable per factory authorisations Allow for adding a GstRTSPAuth on the factory and media level and check permissions when accessing the factory. Add hints to the auth methods for future more fine grained authorisation. Add example application for per factory authentication.
2011-01-12 12:57:09 +00:00
GstRTSPAuth *auth;
rtsp: use AddressPool Remove the multicast_group property. Use the configured addresspool to allocate multicast addresses.
2012-11-14 16:23:59 +00:00
GstRTSPAddressPool*pool;
media: avoid doing _get_state() for state changes When preparing, use the ASYNC_DONE and ERROR messages in the bus handler to wait until the media is prerolled or in error. This avoids doing a blocking call of gst_element_get_state() that can cause lockups when there is an error. Fixes #611899
2010-03-05 16:51:26 +00:00
GstElement *element;
media: add lock to protect state changes
2012-11-13 10:15:35 +00:00
GRecMutex state_lock;
rtsp: massive refactoring Make GObjects from the remaining simple structures. Remove GstRTSPSessionStream, it's not needed. Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how a GstRTSPStream should be transported to a client. Rename GstRTSPMediaFactory::get_element -> create_element because that more accurately describes what it does. Make nice methods instead of poking in the structures. Move some methods inside the relevant object source code. Use GPtrArray to store objects instead of plain arrays, it is more natural and allows us to more easily clean up. Move the allocation of udp ports to the Stream object. The Stream object contains the elements needed to stream the media to a client. Improve the prepare and unprepare methods. Unprepare should now undo everything prepare did. Improve also async unprepare when doing EOS on shutdown. Make sure we always unprepare correctly.
2012-10-25 19:29:58 +00:00
GPtrArray *streams;
media: avoid doing _get_state() for state changes When preparing, use the ASYNC_DONE and ERROR messages in the bus handler to wait until the media is prerolled or in error. This avoids doing a blocking call of gst_element_get_state() that can cause lockups when there is an error. Fixes #611899
2010-03-05 16:51:26 +00:00
GList *dynamic;
GstRTSPMediaStatus status;
rtsp: massive refactoring Make GObjects from the remaining simple structures. Remove GstRTSPSessionStream, it's not needed. Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how a GstRTSPStream should be transported to a client. Rename GstRTSPMediaFactory::get_element -> create_element because that more accurately describes what it does. Make nice methods instead of poking in the structures. Move some methods inside the relevant object source code. Use GPtrArray to store objects instead of plain arrays, it is more natural and allows us to more easily clean up. Move the allocation of udp ports to the Stream object. The Stream object contains the elements needed to stream the media to a client. Improve the prepare and unprepare methods. Unprepare should now undo everything prepare did. Improve also async unprepare when doing EOS on shutdown. Make sure we always unprepare correctly.
2012-10-25 19:29:58 +00:00
gint n_active;
media: ignore spurious ASYNC_DONE messages When we are dynamically adding pads, the addition of the udpsrc elements will trigger an ASYNC_DONE. We have to ignore this because we only want to react to the real ASYNC_DONE when everything is prerolled.
2010-12-11 16:31:44 +00:00
gboolean adding;
Reorganize things, prepare for media sharing Added various other test server examples Move the SDP message generation to a separate helper. Refactor common code for finding the session. Add content-base for realplayer compatibility Clean up request uris before processing for better vlc compatibility. Move prerolling and pipeline construction to the RTSPMedia object. Use multiudpsink for future pipeline reuse.
2009-01-29 12:31:27 +00:00
/* the pipeline for the media */
media: avoid doing _get_state() for state changes When preparing, use the ASYNC_DONE and ERROR messages in the bus handler to wait until the media is prerolled or in error. This avoids doing a blocking call of gst_element_get_state() that can cause lockups when there is an error. Fixes #611899
2010-03-05 16:51:26 +00:00
GstElement *pipeline;
GstElement *fakesink;
GSource *source;
guint id;
Handle media bus messages Handle media bus messages in a custom mainloop and dispatch them to the RTSPMedia objects. Let the default implementation handle some common messages.
2009-02-13 15:39:36 +00:00
media: avoid doing _get_state() for state changes When preparing, use the ASYNC_DONE and ERROR messages in the bus handler to wait until the media is prerolled or in error. This avoids doing a blocking call of gst_element_get_state() that can cause lockups when there is an error. Fixes #611899
2010-03-05 16:51:26 +00:00
gboolean is_live;
media: add a seekable boolean Maintain the seekable state with a new variable instead of reusing the is_live variable.
2011-11-03 11:55:24 +00:00
gboolean seekable;
media: avoid doing _get_state() for state changes When preparing, use the ASYNC_DONE and ERROR messages in the bus handler to wait until the media is prerolled or in error. This avoids doing a blocking call of gst_element_get_state() that can cause lockups when there is an error. Fixes #611899
2010-03-05 16:51:26 +00:00
gboolean buffering;
GstState target_state;
Reorganize things, prepare for media sharing Added various other test server examples Move the SDP message generation to a separate helper. Refactor common code for finding the session. Add content-base for realplayer compatibility Clean up request uris before processing for better vlc compatibility. Move prerolling and pipeline construction to the RTSPMedia object. Use multiudpsink for future pipeline reuse.
2009-01-29 12:31:27 +00:00
/* RTP session manager */
media: avoid doing _get_state() for state changes When preparing, use the ASYNC_DONE and ERROR messages in the bus handler to wait until the media is prerolled or in error. This avoids doing a blocking call of gst_element_get_state() that can cause lockups when there is an error. Fixes #611899
2010-03-05 16:51:26 +00:00
GstElement *rtpbin;
Reorganize things, prepare for media sharing Added various other test server examples Move the SDP message generation to a separate helper. Refactor common code for finding the session. Add content-base for realplayer compatibility Clean up request uris before processing for better vlc compatibility. Move prerolling and pipeline construction to the RTSPMedia object. Use multiudpsink for future pipeline reuse.
2009-01-29 12:31:27 +00:00
Cleanup of sessions and more Fix the refcounting of media and sessions in the client. Properly clean up the session data when the client performs a teardown. Add Server header to responses. Allow for multiple uri setups in one session. Add Range header to the PLAY response and add the range attribute to the SDP message. Fix the session pool remove method, it used the wrong key in the hashtable. Also give the ownership of the sessionid to the session object.
2009-02-04 16:00:42 +00:00
/* the range of media */
media: avoid doing _get_state() for state changes When preparing, use the ASYNC_DONE and ERROR messages in the bus handler to wait until the media is prerolled or in error. This avoids doing a blocking call of gst_element_get_state() that can cause lockups when there is an error. Fixes #611899
2010-03-05 16:51:26 +00:00
GstRTSPTimeRange range;
media: support more Range formats Use the new -base methods to convert the Range string into a seek start and stop value.
2012-11-21 15:41:56 +00:00
GstClockTime range_start;
GstClockTime range_stop;
Initial import
2008-10-09 12:29:12 +00:00
};
Handle media bus messages Handle media bus messages in a custom mainloop and dispatch them to the RTSPMedia objects. Let the default implementation handle some common messages.
2009-02-13 15:39:36 +00:00
/**
* GstRTSPMediaClass:
* @context: the main context for dispatching messages
* @loop: the mainloop for message.
* @thread: the thread dispatching messages.
* @handle_message: handle a message
Added vmethod unprepare to GstRTSPMedia The default implementation sets the state of the pipeline to GST_STATE_NULL
2009-06-12 16:05:30 +00:00
* @unprepare: the default implementation sets the pipeline's state
docs: update docs
2012-10-26 10:33:21 +00:00
* to GST_STATE_NULL and removes all elements.
Handle media bus messages Handle media bus messages in a custom mainloop and dispatch them to the RTSPMedia objects. Let the default implementation handle some common messages.
2009-02-13 15:39:36 +00:00
*
* The RTSP media class
*/
More improvements Rename GstRTSPMediaBin to GstRTSPMedia Parse the request url into a GstRTSPUri object and pass this object to the various handlers and methods that require the uri.
2009-01-22 16:58:19 +00:00
struct _GstRTSPMediaClass {
Initial import
2008-10-09 12:29:12 +00:00
GObjectClass parent_class;
Handle media bus messages Handle media bus messages in a custom mainloop and dispatch them to the RTSPMedia objects. Let the default implementation handle some common messages.
2009-02-13 15:39:36 +00:00
/* thread for the mainloop */
GMainContext *context;
GMainLoop *loop;
GThread *thread;
media: add signal to notify of unprepare
2009-04-03 20:45:57 +00:00
/* vmethods */
rtsp: massive refactoring Make GObjects from the remaining simple structures. Remove GstRTSPSessionStream, it's not needed. Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how a GstRTSPStream should be transported to a client. Rename GstRTSPMediaFactory::get_element -> create_element because that more accurately describes what it does. Make nice methods instead of poking in the structures. Move some methods inside the relevant object source code. Use GPtrArray to store objects instead of plain arrays, it is more natural and allows us to more easily clean up. Move the allocation of udp ports to the Stream object. The Stream object contains the elements needed to stream the media to a client. Improve the prepare and unprepare methods. Unprepare should now undo everything prepare did. Improve also async unprepare when doing EOS on shutdown. Make sure we always unprepare correctly.
2012-10-25 19:29:58 +00:00
gboolean (*handle_message) (GstRTSPMedia *media, GstMessage *message);
gboolean (*unprepare) (GstRTSPMedia *media);
media: add signal to notify of unprepare
2009-04-03 20:45:57 +00:00
/* signals */
media: add signal for new streams This allows applications to listen for new streams and configure properties on them, like the address pool.
2012-11-15 14:41:42 +00:00
gboolean (*new_stream) (GstRTSPMedia *media, GstRTSPStream * stream);
rtsp: massive refactoring Make GObjects from the remaining simple structures. Remove GstRTSPSessionStream, it's not needed. Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how a GstRTSPStream should be transported to a client. Rename GstRTSPMediaFactory::get_element -> create_element because that more accurately describes what it does. Make nice methods instead of poking in the structures. Move some methods inside the relevant object source code. Use GPtrArray to store objects instead of plain arrays, it is more natural and allows us to more easily clean up. Move the allocation of udp ports to the Stream object. The Stream object contains the elements needed to stream the media to a client. Improve the prepare and unprepare methods. Unprepare should now undo everything prepare did. Improve also async unprepare when doing EOS on shutdown. Make sure we always unprepare correctly.
2012-10-25 19:29:58 +00:00
gboolean (*prepared) (GstRTSPMedia *media);
gboolean (*unprepared) (GstRTSPMedia *media);
media: add signal to notify of state changes
2010-12-28 17:34:10 +00:00
rtsp: massive refactoring Make GObjects from the remaining simple structures. Remove GstRTSPSessionStream, it's not needed. Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how a GstRTSPStream should be transported to a client. Rename GstRTSPMediaFactory::get_element -> create_element because that more accurately describes what it does. Make nice methods instead of poking in the structures. Move some methods inside the relevant object source code. Use GPtrArray to store objects instead of plain arrays, it is more natural and allows us to more easily clean up. Move the allocation of udp ports to the Stream object. The Stream object contains the elements needed to stream the media to a client. Improve the prepare and unprepare methods. Unprepare should now undo everything prepare did. Improve also async unprepare when doing EOS on shutdown. Make sure we always unprepare correctly.
2012-10-25 19:29:58 +00:00
gboolean (*new_state) (GstRTSPMedia *media, GstState state);
Initial import
2008-10-09 12:29:12 +00:00
};
More improvements Rename GstRTSPMediaBin to GstRTSPMedia Parse the request url into a GstRTSPUri object and pass this object to the various handlers and methods that require the uri.
2009-01-22 16:58:19 +00:00
GType gst_rtsp_media_get_type (void);
Initial import
2008-10-09 12:29:12 +00:00
Reorganize things, prepare for media sharing Added various other test server examples Move the SDP message generation to a separate helper. Refactor common code for finding the session. Add content-base for realplayer compatibility Clean up request uris before processing for better vlc compatibility. Move prerolling and pipeline construction to the RTSPMedia object. Use multiudpsink for future pipeline reuse.
2009-01-29 12:31:27 +00:00
/* creating the media */
GstRTSPMedia * gst_rtsp_media_new (void);
Add shared properties to media and factory Add the shared property to media. Implement some simple caching in the factory depending on if the media is shared or not.
2009-01-29 16:20:27 +00:00
void gst_rtsp_media_set_shared (GstRTSPMedia *media, gboolean shared);
gboolean gst_rtsp_media_is_shared (GstRTSPMedia *media);
Initial import
2008-10-09 12:29:12 +00:00
media: more work on making the media shared Add a reusable flag to medias, indicating that they can be reused after a state change to NULL. Small cleanups.
2009-04-03 20:22:30 +00:00
void gst_rtsp_media_set_reusable (GstRTSPMedia *media, gboolean reusable);
gboolean gst_rtsp_media_is_reusable (GstRTSPMedia *media);
media: allow configuration of allowed lower transport
2010-03-19 14:15:29 +00:00
void gst_rtsp_media_set_protocols (GstRTSPMedia *media, GstRTSPLowerTrans protocols);
GstRTSPLowerTrans gst_rtsp_media_get_protocols (GstRTSPMedia *media);
media-factory: add eos-shutdown property Add an eos-shutdown property that will send an EOS to the pipeline before shutting it down. This allows for nice cleanup in case of a muxer. Fixes #625597
2010-08-20 16:17:08 +00:00
void gst_rtsp_media_set_eos_shutdown (GstRTSPMedia *media, gboolean eos_shutdown);
gboolean gst_rtsp_media_is_eos_shutdown (GstRTSPMedia *media);
media: more work on making the media shared Add a reusable flag to medias, indicating that they can be reused after a state change to NULL. Small cleanups.
2009-04-03 20:22:30 +00:00
media: enable per factory authorisations Allow for adding a GstRTSPAuth on the factory and media level and check permissions when accessing the factory. Add hints to the auth methods for future more fine grained authorisation. Add example application for per factory authentication.
2011-01-12 12:57:09 +00:00
void gst_rtsp_media_set_auth (GstRTSPMedia *media, GstRTSPAuth *auth);
GstRTSPAuth * gst_rtsp_media_get_auth (GstRTSPMedia *media);
rtsp: use AddressPool Remove the multicast_group property. Use the configured addresspool to allocate multicast addresses.
2012-11-14 16:23:59 +00:00
void gst_rtsp_media_set_address_pool (GstRTSPMedia *media, GstRTSPAddressPool *pool);
GstRTSPAddressPool * gst_rtsp_media_get_address_pool (GstRTSPMedia *media);
media: add property to configure kernel buffer sizes Add a property to configure the kernel UDP buffer size.
2011-01-31 16:28:22 +00:00
void gst_rtsp_media_set_buffer_size (GstRTSPMedia *media, guint size);
guint gst_rtsp_media_get_buffer_size (GstRTSPMedia *media);
media: add property for multicast group Add a property to configure the multicast group in the media. Based on patches from Marc Leeman and Robert Krakora.
2011-08-16 11:25:16 +00:00
Reorganize things, prepare for media sharing Added various other test server examples Move the SDP message generation to a separate helper. Refactor common code for finding the session. Add content-base for realplayer compatibility Clean up request uris before processing for better vlc compatibility. Move prerolling and pipeline construction to the RTSPMedia object. Use multiudpsink for future pipeline reuse.
2009-01-29 12:31:27 +00:00
/* prepare the media for playback */
gboolean gst_rtsp_media_prepare (GstRTSPMedia *media);
media: more work on making the media shared Add a reusable flag to medias, indicating that they can be reused after a state change to NULL. Small cleanups.
2009-04-03 20:22:30 +00:00
gboolean gst_rtsp_media_unprepare (GstRTSPMedia *media);
Reorganize things, prepare for media sharing Added various other test server examples Move the SDP message generation to a separate helper. Refactor common code for finding the session. Add content-base for realplayer compatibility Clean up request uris before processing for better vlc compatibility. Move prerolling and pipeline construction to the RTSPMedia object. Use multiudpsink for future pipeline reuse.
2009-01-29 12:31:27 +00:00
rtsp: massive refactoring Make GObjects from the remaining simple structures. Remove GstRTSPSessionStream, it's not needed. Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how a GstRTSPStream should be transported to a client. Rename GstRTSPMediaFactory::get_element -> create_element because that more accurately describes what it does. Make nice methods instead of poking in the structures. Move some methods inside the relevant object source code. Use GPtrArray to store objects instead of plain arrays, it is more natural and allows us to more easily clean up. Move the allocation of udp ports to the Stream object. The Stream object contains the elements needed to stream the media to a client. Improve the prepare and unprepare methods. Unprepare should now undo everything prepare did. Improve also async unprepare when doing EOS on shutdown. Make sure we always unprepare correctly.
2012-10-25 19:29:58 +00:00
/* creating streams */
void gst_rtsp_media_collect_streams (GstRTSPMedia *media);
GstRTSPStream * gst_rtsp_media_create_stream (GstRTSPMedia *media,
GstElement *payloader,
GstPad *srcpad);
Add shared properties to media and factory Add the shared property to media. Implement some simple caching in the factory depending on if the media is shared or not.
2009-01-29 16:20:27 +00:00
/* dealing with the media */
guint gst_rtsp_media_n_streams (GstRTSPMedia *media);
rtsp: massive refactoring Make GObjects from the remaining simple structures. Remove GstRTSPSessionStream, it's not needed. Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how a GstRTSPStream should be transported to a client. Rename GstRTSPMediaFactory::get_element -> create_element because that more accurately describes what it does. Make nice methods instead of poking in the structures. Move some methods inside the relevant object source code. Use GPtrArray to store objects instead of plain arrays, it is more natural and allows us to more easily clean up. Move the allocation of udp ports to the Stream object. The Stream object contains the elements needed to stream the media to a client. Improve the prepare and unprepare methods. Unprepare should now undo everything prepare did. Improve also async unprepare when doing EOS on shutdown. Make sure we always unprepare correctly.
2012-10-25 19:29:58 +00:00
GstRTSPStream * gst_rtsp_media_get_stream (GstRTSPMedia *media, guint idx);
Add shared properties to media and factory Add the shared property to media. Implement some simple caching in the factory depending on if the media is shared or not.
2009-01-29 16:20:27 +00:00
Add beginnings of seeking. Parse the Range header and perform a seek on the pipeline for the requested position. It's disabled currently until I figure out what's going wrong.
2009-03-12 19:32:14 +00:00
gboolean gst_rtsp_media_seek (GstRTSPMedia *media, GstRTSPTimeRange *range);
media: make method to retrieve the play range Make a method to retrieve the playback range so that we can conditionally create a different range for the SDP and the PLAY requests.
2010-12-28 17:35:01 +00:00
gchar * gst_rtsp_media_get_range_string (GstRTSPMedia *media, gboolean play);
Add beginnings of seeking. Parse the Range header and perform a seek on the pipeline for the requested position. It's disabled currently until I figure out what's going wrong.
2009-03-12 19:32:14 +00:00
rtsp: massive refactoring Make GObjects from the remaining simple structures. Remove GstRTSPSessionStream, it's not needed. Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how a GstRTSPStream should be transported to a client. Rename GstRTSPMediaFactory::get_element -> create_element because that more accurately describes what it does. Make nice methods instead of poking in the structures. Move some methods inside the relevant object source code. Use GPtrArray to store objects instead of plain arrays, it is more natural and allows us to more easily clean up. Move the allocation of udp ports to the Stream object. The Stream object contains the elements needed to stream the media to a client. Improve the prepare and unprepare methods. Unprepare should now undo everything prepare did. Improve also async unprepare when doing EOS on shutdown. Make sure we always unprepare correctly.
2012-10-25 19:29:58 +00:00
gboolean gst_rtsp_media_set_state (GstRTSPMedia *media, GstState state,
GPtrArray *transports);
Added gst_rtsp_media_remove_elements function
2009-06-13 12:38:20 +00:00
Initial import
2008-10-09 12:29:12 +00:00
G_END_DECLS
#endif /* __GST_RTSP_MEDIA_H__ */
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